The present invention relates generally to a composite switched network system capable of reducing costs for equipment and operations by integrating a circuit switched network such as a telephone network with a connectionless packet switched network such as an Internet network or an Intranet network, and more particularly to an Internet telephone system capable of communications of voice information in the form of packets via the Internet network or the Intranet network.
With a progress of networking technology such as acquiring a much broader band of Local Area Network (LAN) and a progress of PC technology such as attaining multi-functions of a personal computer (PC) and a speed-up of a CPU applied to the PC over the recent years, it is feasible in terms of utilization to perform communications of voice information at a high speed between the PCs on a plurality of LANs. With those technological progresses, application software executed between the PCs on the Internet network (which includes the Intranet network if not particularly specified in this Specification), constructed of private lines for voice communications on conventional telephones, the LAN and Wide Area Network (WAN) and a hardware system incorporating this software, have been rapidly put into markets. This system is called [Internet telephony].
In this Internet telephony system, a VoIP (Voice over Internet Protocol) technology is used for integrating the voice communications with data communications. The VoIP technology is that the voice is converted into a frame at an interval of a short time (on the order of 20 ms) according to a network layer protocol used in the Internet, i.e., on the network having the IP (IP network), and a packet assembled by adding an IP header to the frame is transmitted and received. Note that the communications of the voice information in a computer network have a long history in which the first packet switching was experimented in 1974 in ARPANET (Advanced Research Projects Agency Network) as the predecessor to the Internet (refer to “RFC741: Specifications for the Network Voice Protocol (NVP)”, IETF, Danny Cohen, 1976).
Further, there has been developed an Internet telephony gateway which incorporates a gateway functions for making a communications protocol conversion between the telephone network and the Internet network and actualizes the communications between the telephone network and the Internet network. There was proposed and has already been utilized a system (Internet-telephony-based cored system) in which the information is relayed via the Internet network between respective switches in the conventional telephone network by use of the Internet telephony gateways.
Generally, an operation for a telephone talk in the cored system of the Internet telephony has a lower cost than a telephone talk (voice communication) through the conventional telephone network, and therefore this cored system and services are now in the process of a rapidly spread. The transmission (transfer) standard in the cored system using the Internet telephony gateways is not, however, established, and what exists at the present is only ITU-T Recommendation H.450.2 (H.323) which defines the services within the Internet network, and the services in the conventional telephone network. Namely, there is no transmission standard on which to communicate the information to between the Internet telephony gateway having a function of converting the voice information given from the telephone network into a packet on the basis of the Internet Protocol (IP) and a router incorporating a function of routing the IP packet.
In the Internet telephony system, it is indispensable for routing the voice packet based on the Internet Protocol while restraining a delay to fragment short each of data packets (such as file-transfer packets) flowing simultaneously in the Internet network and thus transfer those packets. To describe it in depth, when starting the forwarding of the data packet as in the file transfer even if the voice packet is controlled with a high priority, the voice packet exhibiting a higher urgency which has arrived at afterward can not be forwarded if not waiting till a transmission of the data packet now undergoing the processing is finished. In a low-speed link, this waiting time is long, and a voice quality declines due to a transfer delay of the voice packet. In the case of transferring, for example, a 1500-byte packet onto a private line having a transmission speed on the order of 64 Kbps, a time of 180 ms is needed. A time for which the voice delay is allowable in the telephone talk is, generally speaking, 200 ms between the terminals (End-to-End), and the waiting time of 180 ms described above is extremely long.
A scheme for obviating this problem is to reduce a packet processing time by fragmenting the data packet having a packet length into pieces of small fragments and transfer the packet in such a manner that the voice packet is inserted into a gap between the fragmented data packets, thereby restraining the transfer delay. A prior art for thus fragmenting the data packet may involve multilink PPP (Point-to-Point Protocol) interleaving (IETF (Internet Engineering Task Force) draft: MCML) and setting an MTU length. According to this prior art, however, the data packet is always fragmented even when the voice packet does not flow, and hence a decrease in efficiency is inevitable in terms of transferring the data packet. That is, in the case of the data packet having a short packet length, the router as a relay device in the network has a high load. As a result, for instance, a file transfer is finished sooner by forwarding the data packet on a 1500-byte basis than forwarding the data packet with a fragmentation by every 300 bytes.